Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
2010/12/21 Paul Belanger : > On 10-12-20 05:51 PM, Jarek Jarzebowski wrote: >> OK, so I have attached debug log. >> >> I am using: >> *CLI> core show version >> Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on >> 2010-12-17 23:03:58 UTC >> > Definitely a bug, ran into the same issue with chan_iax2 and DNS > lookups. Please open a new issue on the tracker, include your debug log > and sip.conf. So I have opened new issue #0018514. > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Regards, Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
On 10-12-20 05:51 PM, Jarek Jarzebowski wrote: > OK, so I have attached debug log. > > I am using: > *CLI> core show version > Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on > 2010-12-17 23:03:58 UTC > Definitely a bug, ran into the same issue with chan_iax2 and DNS lookups. Please open a new issue on the tracker, include your debug log and sip.conf. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
2010/12/20 Paul Belanger : > On 10-12-20 04:41 AM, Jarek Jarzebowski wrote: >> [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len >> 843) to 0.0.4.26:5060 returned -1: Invalid argument >> > It looks to be a regression with the IPv6 code added to chan_sip. Which > version of 1.8 are you using? I'd also be good to see a full debug[1] log. OK, so I have attached debug log. I am using: *CLI> core show version Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on 2010-12-17 23:03:58 UTC > > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Regards, Jarek full.tgz Description: GNU Zip compressed data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
2010/12/20 Jeremy Kister : > On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote: >> >> Now, when I Dial extension 1050, and there is no 1050 peer registered I >> got: >> >> [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len >> 843) to 0.0.4.26:5060 returned -1: Invalid argument > > You haven't done anything wrong; I have the same issue. > > Just add it to the list of things to fix in 1.8.. > > Do you want to add it to http://issues.asterisk.org ? Yes, of course. I want to add it to http://issues.asterisk.org > > -- > > Jeremy Kister > http://jeremy.kister.net./ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
On 10-12-20 04:41 AM, Jarek Jarzebowski wrote: > [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len > 843) to 0.0.4.26:5060 returned -1: Invalid argument > It looks to be a regression with the IPv6 code added to chan_sip. Which version of 1.8 are you using? I'd also be good to see a full debug[1] log. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote: Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument You haven't done anything wrong; I have the same issue. Just add it to the list of things to fix in 1.8.. Do you want to add it to http://issues.asterisk.org ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
Hi All,
I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend.
My dialplan:
exten => _,1,Dial(SIP/${EXTEN},60,rt)
Now, when I Dial extension 1050, and there is no 1050 peer registered I got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
In 1.6 there was no problem, I have got Channel is UNAVAILABLE message
and hangup.
What have I missed in 1.8?
Regards,
Jarek
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